Don't understand little things (don't undestand anythings about digital ) and have please just two questions:

if the DLCP is a preamplifier too, where is the volume control ? On the software...on the computer which must be switched on all the time) ? (if a cd player or an individual steramer is connected via spdif). in a case of a streamer like a squeezebox duet used with spdif output: can I use the volume controle of the streamer without any loss ?

Understand that Hypex is not full "digital" like the little FDA amp of HifiMeDiy DDX320 with its little remote ant its 14 tone curves (of course I don't want here to compare these two products which are not made for the same people). But if the DLCP is made in relation to the Ncore amp first and for the others brand after, is there possibility to make shorter between the DLCPs' DACs--> OPA1632 and the "analogic" input stage of the hypex amp and its amp stage ?

Sorry for the low technic level questions, not sure to understand the different topologies about digital amplifier...

Coefficient length is rarely an issue. Coefficient rounding only causes minor errors in the frequency response which are only noticeable when you make very high Q's at low frequencies. 28 bit coefficients are more than enough for any use sensibly related to loudspeakers. What really matters sonically is the length of the data words. Data rounding translates directly as distortion. In IIR filters you very often find yourself subtracting two nearly identical numbers and then amplifying the rest. Single precision floating point is slightly better than 24 bits fixed because at the very least the error scales with the signal level. But then again for most practical purposes 32-bit fixed is always better than single precision float - float only wins when the signal level drops below -48dB.

The data word lenth in the TAS3108 is 56 bits (48 after the point) and its accumulator is even longer. That's the main reason why I love it so. It's also got a cute microcoded architecture. You can seriously geek out writing code for this chip. I'm seriously galled that the 3108 has been relegated to "not recommended for new design" with no alternative in sight. The "alternative" given on the TI site is ridiculous. What this means is that we'll be doing a lifetime buy when they announce end of production.

The SigmaDSP chips have a similar microcoded architecture but its word length is limited to 28 bits (data and coeff), with limited support for double precision. If you want to do double precision you have to manipulate half words yourself which doubles the number of cycles needed. They're no alternative for the 3108 but their built-in converters and more flexible I/O give them their proper niche.

For those still believing that class D should be closer to digital than a linear amplifier, please read http://www.hypex.nl/docs/papers/AES120_353BP.pdf. The notion of "digital amplifiers" is nonsensical. The only thing you can conceivably eliminate between a DAC and an Ncore amp is the input buffer stage if you change the gain of the DAC's post filter. That would mean a different filter for each prospective DLCP/amp pairing, which is not economical.

As amply attested, I only believe in FIR stages as a way to approximate poles outside the unit circle targeted specifically to cancel non-minimum phase zeros that inevitably result from summing higher order LPF/HPF slopes. There simply aren't justifiable uses of FIR otherwise. Some future update of the software and firmware will support FIR approximation of noncausal IIR. You don't need a lot of DSP for that. The LS1 has a 6000 tap filter correcting the 70Hz subwoofer crossover, and it's done in a hundred cycles or so. The main question will be how to make the software intelligently trade delay memory for this use and for time alignment. We know how to do it, just not whose time to do it in.

The LS1 has a 6000 tap filter correcting the 70Hz subwoofer crossover, and it's done in a hundred cycles or so. The main question will be how to make the software intelligently trade delay memory for this use and for time alignment. We know how to do it, just not whose time to do it in.

Back a while ago you stated that linear phase was overrated and that the advantage of using the FIR correctional filter in the LS1 didn't affect sound much at all. You theorized though that it would be more audible at lower frequencies.

So, now that you've designed the LS1s, did you find that this prediction was correct? Was FIR correction was more of an advantage lower down or is it hard to detect the improvement there also?

You should get much more audible results with the phase linearization of this 70hz crossover than the ones you got with the 1.5khz crossover, according to the majority of reports I have from users of rePhase.
(by the way Bruno, this was your LS1 white paper that gave me the motivation to write this software )